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Original commit message from CVS: * ext/audiofile/gstafsrc.c: (gst_afsrc_get): Remove old debug output * ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type), (gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state), (gst_dvdec_set_property), (gst_dvdec_get_property): Change the quality setting to an enum, so it works from gst-launch Don't renegotiate a non-linked pad. Allows audio only decoding. * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps), (gst_deinterlace_link), (gst_deinterlace_init): * gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps), (gst_videodrop_link): Some caps negotiation fixes |
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gstaf.c | ||
gstafparse.c | ||
gstafparse.h | ||
gstafsink.c | ||
gstafsink.h | ||
gstafsrc.c | ||
gstafsrc.h | ||
Makefile.am | ||
README |
This plugin wraps the SGI Audiofile (http://oss.sgi.com/projects/audiofile/) library into a src and sink element. You can read from and write to the supported formats (WAVE, AIFF, AIFFC, NEXTSND). What is supported : * all the file formats * integer sample data, both 2's complement and unsigned * 8 or 16 bit width & depth (haven't tested others) * sample rate * some sort of endianness control What isn't supported yet : * float data What you can do : * src element only accepts location argument * sink element accepts location, endianness and type - location : file on the system to output - endianness : at this time endianness is still a bit shady you can either set 1234 or 4321; setting it to 4321 will byteswap the buffer data you might want to keep it at 1234 for now - type : one of the file types Use gstreamer-inspect on afsink and afsrc to see all of the supported options. Examples : * tools/gstreamer-launch afsrc location=/opt/media/wav/dark-480-16-m.wav ! afsink type=2 location=/opt/media/wav/dark-480-16-m.aiff Future plans : * add float support * wrap up afsink and afsrc with pipe and fork to act like data convertors, allowing arbitrary choice of sink and src element